July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

get answertime dialstatus after bridge
by salzh 03 Jul '11

03 Jul '11

hi. is there any way to get a call information such as answertime dialstatus and so on after executing 'bridge' application? just like what the application 'dial' do in asterisk.

1 0

how to change conference's caller-id-number at run-time?
by ran zhang 02 Jul '11

02 Jul '11

I want when the bridging conference is inviting someone to start the conference, the caller is shown as the real caller id number rather than whats stored in 'caller-id-number' in conference.conf.xml.

1 0

Inband DTMF set in Event Socket
by rex.alex 02 Jul '11

02 Jul '11

Hello, How to set the DTMF type as inband when I dial through event socket using api originate command? Thanks, Rex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inband-DTMF-set-in-Event-Sock… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 7

LUA Script: IGNORE INFO DTMF
by ankIT WALiA 02 Jul '11

02 Jul '11

Hi, I am trying to get the digits pressed using statement in my LUA script "digits = session:playAndGetDigits(3, 5, 3, 7000, "#", prompt, invalid, "\\d+")" The FS is not able to take the digits and giving the invalid response always. 2011-07-02 15:51:30.405443 [WARNING] sofia.c:6715 IGNORE INFO DTMF(7) (This channel was not configured to use INFO DTMF!) I am trying establish a channel between default 1002 ext and custom ext. Am I missing something? Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

2 2

Linksys PAP2 tls with freeswitch
by Mohammad Emran 02 Jul '11

02 Jul '11

Hi all, Please help me to configure linksys pap2 tls with freeswitch.i have configured freeswitch as per wiki. Thank you. Sent from my iPad

1 0

REGISTER response sent to the wrong por
by Dave Horton 02 Jul '11

02 Jul '11

I have a phone behind a firewall registering with the internal profile. The address in the Via header is not reachable, and the FS (correctly, IMO) sends the response back to the actual sending IP address. However, rather than sending it to the actual sending udp port, it incorrectly (again, IMO) sends it to the port advertised in the Via header. Of course the message does not get through the firewall, and the phone keeps retransmitting the REGISTER. Is this a bug, or am I missing something? And is there a setting where I can force the behavior that I want (which is: send responses to the sending address:port if the sending address:port is different than the Via header). Trace below.. recv 481 bytes from udp/[66.37.97.53]:1024 at 19:28:12.445081: ------------------------------------------------------------------------ REGISTER sip:65.162.239.204 SIP/2.0 Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf From: "1002" <sip:1002@65.162.239.204>;tag=7718f2af7f3dcc73o2 To: "1002" <sip:1002@65.162.239.204> Call-ID: 930c7e31-648ae3db(a)10.93.14.119 CSeq: 53622 REGISTER Max-Forwards: 70 Contact: "1002" <sip:1002@10.93.14.119:5064>;expires=3600 User-Agent: Linksys/SPA962-5.2.8(SC) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces ------------------------------------------------------------------------ send 671 bytes to udp/[66.37.97.53]:5064 at 19:28:12.445259: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf;received=66.37.97.53 From: "1002" <sip:1002@65.162.239.204>;tag=7718f2af7f3dcc73o2 To: "1002" <sip:1002@65.162.239.204>;tag=t1m9emDKB4trF Call-ID: 930c7e31-648ae3db(a)10.93.14.119 CSeq: 53622 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-25c725c 2011-06-30 18-30-24 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="65.162.239.204", nonce="7e426985-2f7a-40ff-a817-6c167207853f", algorithm=MD5, qop="auth" Content-Length: 0

4 3

change sounds file format of voicemail
by Madovsky 02 Jul '11

02 Jul '11

Where can I force voicemail to use sounds other than .wav ? Thanks

4 7

Could not connect thru GSMOpen
by ankIT WALiA 01 Jul '11

01 Jul '11

Hi all, I have properly compiled the gsmopen module. Added the Dialplan <extension name="gsmopen"> <condition field="destination_number" expression="^1002$"> <action application="bridge" data="gsmopen/interface1/xxxxxxxxx"/> </condition> </extension> ---------------------------------------------------- And the gsm conf file <configuration name="gsmopen.conf" description="GSMopen Configuration"> - <global_settings> <param name="debug" value="8"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="hold-music" value="$${moh_uri}"/> <param name="destination" value="9999"/> </global_settings>  - <per_interface_settings> - <interface id="1" name="interface1"> <param name="hold-music" value="$${moh_uri}"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="destination" value="5000"/> <param name="alsacname" value="plughw:1"/> <param name="alsapname" value="plughw:1"/> <param name="controldevice_name" value="/dev/ttyACM1"/> </interface>  </per_interface_settings> </configuration> ------------------------------------------------------------------ I am using a Nokia handset in India --------------------------------------------------------------------- I am getting the below error on gsm_load. gsm reload 2011-06-30 19:03:52.716491 [WARNING] mod_gsmopen.cpp:1860 rev exported[(nil)|37 ][WARNINGA 1860 ][interface1][-1, 0, 0] STARTING interface_id=1 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2999 rev exported[(nil)|37 ][ERRORA 2999 ][interface1][-1, 0, 0] snd_pcm_open failed with error 'No such file or directory' on device 'plughw:1', if you are using a plughw:n device please change it to be a default:n device (so to allow it to be shared with other concurrent programs), or maybe you are using an ALSA voicemodem and slmodemd is running? 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2891 rev exported[(nil)|37 ][ERRORA 2891 ][interface1][-1, 0, 0] Failed opening ALSA capture device: plughw:1 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1931 rev exported[(nil)|37 ][ERRORA 1931 ][interface1][-1, 0, 0] alsa_init failed 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1932 rev exported[(nil)|37 ][ERRORA 1932 ][interface1][-1, 0, 0] STARTING interface_id=1 FAILED 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:3146 rev exported[(nil)|37 ][ERRORA 3146 ][interface1][-1, 0, 0] ALARM on interface interface1: ----------------------------------------------------------------------------------------------------------------- I could not understand what does the error message trying to say? I am connected to friend's mobile with datacable on com1 port. I jsut wanted to check SMS. Please help. Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

3 3

Generating ringback if 180 arrives after 183
by Matteo 01 Jul '11

01 Jul '11

Hi all, I've the following scenario from my upstream provider : FS calls provider Provider sends easrly media with 183 announcing that the call is going to be trasferred Provider sends 180 without sdp In my dialplan I'm using ringback variable to let FS generate ringback to internal network, and if I receive 180 without 183 is working ok. What I'm trying to achieve is to let ringback work even if we've already received the 183 with sdp. I've tryed a lot of dialplan magic without luck. Maybe we need to add this support into FS itself ? What I see seems that we need to break the bridge or at least "inject" the tone stream into the internal leg of the call. To be fair, the exact sequence is: FS calls provider provider sends 183/sdp annoucing the call trasnfer provider sends update which stops the media (media=inactive) provider sends 180 ringing (we must generate here) maybe the correct way is to handle the update, put fs back into ringing state and proceed from there? Is feasible or we can get the same result with some dialplan magic? any hint is appreciated :) regards, matteo

2 5

How to send Outbound Caller ID Name out Sangoma A101 PRI
by Robert Hadley 01 Jul '11

01 Jul '11

Hi FS Users, Is there a way to send the outbound caller ID name when using FreeTDM, wanpipe, and a Sangoma A101 PRI? I am setting the effective_caller_id_name and _number and the number goes out but not the name. I also tried setting the origination_caller_id_name on the bridge but that didn't work either. I receive the incoming caller_id_name in a separate Facility message. Thanks, Robert

4 3